The difference "should" be inaudible, yet in A/B (not ABX) tests, the preference for these inaudible filters matches the preference for audible filters demonstrated in this thread - closer to minimum than linear phase; more gentle than brick wall.

A/B tests, you mean sighted evaluations?

We can't rely on just gentle slope filters because we need notches to get rid of hum and other similar interfering signals.

QUOTE

So, do we have the courage (after 3 pages of warming up) to design a test to find out if >20kHz brick wall filter artifacts can be audible ?

44 Khz sample rate audio is endemic. It is almost totally based on > 20 KHz brick wall filters. Most good implmentations are sonically transparent, that is they defy detection in ABX tests.

QUOTE

I'd be surprised if tests like this have not been done before, so perhaps we can learn from them.

This is the earliest test of > 20 KHz brick wall filtering that I know of - late 1970s, or very early 1980s:

If CEP (mentioned in 1st post) means Cool Edit Pro and, if CEP supports VST plug-ins and if you plan to record the filtered output then, there's a plug-in called RubberFilter made by Christian-W. Budde. Rubberfilter provides up to 64th order Butterworth filters to ‘rubber’ out frequencies.

If CEP does not support recording the FX output then, at least Coscos Reaper supports this feature.

If a VST implementation is OK for you then, there's also Butterworth and Cebyshev HP/LP modules for Delphi and SynthEdit available (same site). SynthEdit is easy to use VST creation environment ... it took less than half of an hour for me to build a "twin" HP filter type of 1-64th order Butterworth/1-32th order Cebyshev.

There's some - not very scientific - evidence that steep anti-aliasing filters degrade certain subtle aspects of audio quality, although ringing takes place in the ultrasound range: "Effects in High Sample Rate Audio Material".

it seems that these were sighted comparisons. Although an interesting attempt, and correlating with many professional engineers' opinions, it's not an acceptable proof in this forum's sense. The differences attributed to the filters could (QED) completely disappear in a double blind test.Do you have any plans to evaluate filter effects (e.g. ringing) for src.infinitewave.ca ?Since most DACs are linear phase and therefore suffer from ringing effects themselves, what would be a good setup to test for audible effect ? (my guess: upsampling to as high as the DAC allows)

There's some - not very scientific - evidence that steep anti-aliasing filters degrade certain subtle aspects of audio quality, although ringing takes place in the ultrasound range: "Effects in High Sample Rate Audio Material".

IMO, the key text from that piece which I first downloaded from the DCS web site is:

"Recording engineers, and many musicians – particularly in the classical area – are becoming awarethat material recorded and edited using these higher sample rates has some attractive qualities.Current theory on how human hearing works has so far been unable to explain the basis for thesequalities, but they are none the less easy to demonstrate."

The key phrase is "...they are none the less easy to demonstrate". Anybody who has spent as much time and money as I have spent out of my own personal pocket (and the pockets of my friends), in sincere and sell-supported attempts to demonstrate the possible benefits of increased bandpass on sonic accuracy, can't avoid having a bit of an emotional reaction.

I translate the quoted paragraph above as reading "Long-established science says that sample rates higher than 44.1 KHz have no audible benefts, but we're from DCS and our business plan is based on selling overpriced DACs for fun and profit, so we're going to throw stones at established science and see if enough people buy into our little anti-scientific song and dance to make us rich."

At this point, the audio industry had its little fling with SACD and DVD-A, and it is now pretty clear that at least some of the time, you can't fool enough people to make a good mainstream business out of fooling them.

QUOTE (Kees de Visser @ Dec 23 2008, 07:53)

Hi Alex,

it seems that these were sighted comparisons. Although an interesting attempt, and correlating with many professional engineers' opinions, it's not an acceptable proof in this forum's sense. The differences attributed to the filters could (QED) completely disappear in a double blind test.Do you have any plans to evaluate filter effects (e.g. ringing) for src.infinitewave.ca ?Since most DACs are linear phase and therefore suffer from ringing effects themselves, what would be a good setup to test for audible effect ? (my guess: upsampling to as high as the DAC allows)

I've participated in a number of attempts to show the non-transparency of various brick wall filters operating with a 22 KHz bandpass.

The first time was a few years before the CD came on the market, probably 1979 or 80. Our source material was live studio sound and 15 ips half track masters. The filters in question were part of a digital delay that Ampex was marketing to people who had automated cutting lathes. I suspect the brick wall filters were implemented in the analog domain, and were closest to being minimum phase.

Later on I did a bunch of demos for the now-defunct PCABX web site. The origional source material was recorded @24/96 using B&K 4006 measurement mics. The brick wall filters were implemented with CoolEdit's FFT filter.

IMO, the interesting question is not whether 22 KHz brick wall filters are sonically transparent, but rather how low can you push brick wall filters before they start being noticable.

IME with most music (one exception being KikeG's little ca. 16 KHz resonator) a 16 KHz brick wall low pass can be pretty innocent. It seems like a lot of people building perceptual coders tend to agree that < 22 Khz is OK.

I agree with you. I never had the chance to hear the differences myself. However many people claim that they do hear differences of different SRC algorithms, for example, depending on the filter type used (again all above 20 kHz).

I agree with you. I never had the chance to hear the differences myself. However many people claim that they do hear differences of different SRC algorithms, for example, depending on the filter type used (again all above 20 kHz).

I question some of the means that people use to hear differences related to different SRC algorithms. One approach is to take some regular music in say the 24 bit domain, attenuate it by 40 dB, downsample it to 16 bits, amplify it by 40 dB, and then present the results for comparison.

Do you think that is representative of how people actually use SRC software?

SRC converts sampling rate, not the bit depth. However your test looks good for testing of bit depth reduction. One important point is that listening levels are adjusted so that the dithering noise is kept inaudible or barely audible in this test - just as it is in most real situations.

Right, but SRC is a generic term, used for software that changes both wordlength and sample rate.

QUOTE

However your test looks good for testing of bit depth reduction.

Really, do you seriously think that is how people use SRC software in day-to-day applications?

QUOTE

One important point is that listening levels are adjusted so that the dithering noise is kept inaudible or barely audible in this test - just as it is in most real situations.

Real situations where dither noise is audible? I can't think of one situation where the background noise on a CD I was listening to was related to dither. Room noise always dominates in real world recordings, doesn't it?

So, do we have the courage (after 3 pages of warming up) to design a test to find out if >20kHz brick wall filter artifacts can be audible ?I'd be surprised if tests like this have not been done before, so perhaps we can learn from them.

Maybe "bandpass" would be happy to re-run the filter generation he did on page 2 of this thread, but with a sampling rate of 96kHz, and a low pass filter cut-off frequency of 20kHz - giving us minimum phase, maximum phase, linear phase (050, as previously labelled), and (for the sake of it) 030 and 070.

The resulting impulses can be listened to as-is, or convolved with 96kHz material to implement the appropriate filtering. Note convolution normally involves time domain reversal, so minimum and maximum phase will swap - unless the particular convolution implementation explicitly re-reverses the filter coefficients.

So, do we have the courage (after 3 pages of warming up) to design a test to find out if >20kHz brick wall filter artifacts can be audible ?I'd be surprised if tests like this have not been done before, so perhaps we can learn from them.

Maybe "bandpass" would be happy to re-run the filter generation he did on page 2 of this thread, but with a sampling rate of 96kHz, and a low pass filter cut-off frequency of 20kHz - giving us minimum phase, maximum phase, linear phase (050, as previously labelled), and (for the sake of it) 030 and 070.

The resulting impulses can be listened to as-is, or convolved with 96kHz material to implement the appropriate filtering. Note convolution normally involves time domain reversal, so minimum and maximum phase will swap - unless the particular convolution implementation explicitly re-reverses the filter coefficients.

If you're near London, you might be interested to hear tomorrow's AES lecture / interview (it's a "Christmas special"!), which I believe may include discussion of the choice of minimum phase filters instead of linear phase filters for anti-image filtering. I assume this is to reduce pre-ringing, though do not understand how this could be audible at 22kHz!

Here are the (selective) notes I made on the night. (re-listening to the mp3 might reveal they're inaccurate, but I don't have two hours to spare!)

I didn't note anything that I was already familiar with. Everything below (unless marked) is my recollection of his opinion, and not my own.

It's a report of a lecture, not a statement of fact. No TOS strikes please!

2008-12-09 AES UK Bob Stuart Meridian

Active Loudspeakers. He doesn't like correcting response errors narrower than the auditory filter bandwidth. It's amazing what you can hear through distortion - e.g. you might have 10% speaker distortion, 0.01% amplifier distortion, and you can still hear the error in the 22nd bit of a filter.

Room correction. He think this is safe below ~200-300Hz, but treacherous above that. For one thing, it smears the time domain. For another, he believes speakers should be in the same room as the listener. We don't equalise real people talking in a room to correct for the room!

DVD-A. He wanted to give something back to the industry that was worthwhile, and better than CD. Before DVD-A he was involved with Xtrabits, the work for which revealed the required noise floor. There's a paper: Coding for High Res audio systems, AES, March 2004.

He discussed the required noise floor using several frequency/amplitude plots showing the threshold of hearing (lower curve), the maximum peak replay level (straight line at the top, he set this to 114dB SPL), and the audible distortion or noise of various systems.

16-bit TPDF dither just exceeds the threshold of hearing in the most sensitive region at this playback level. Re-dither or re-quantisation can be very audible.

At 20-bits, it's OK, and you can do 5 processes before it touches threshold of hearing.

24-bits: "ludicrously over specified".

He mentioned some room noise data from Louis Fielder that's relevant to this discussion.

Then he worked through an interesting set of statements which explain why CD is nearly good enough:

Firstly, the noise in even high-res recording is always above that which can be achieved on CD.

Secondly, while it's true that 120dB at 24kHz can't be stored on CD (despite being audible to some people), there's no actual content like that, and it would melt tweeters anyway. A "real world" worst case is something like a cymbal crash, and by about 20kHz even that is below the threshold of hearing.

The preference for higher sample rates is not based on ultrasonic content, but on time resolution and aliasing when conversion goes wrong. All anti-alias filters are brick wall, anti-causal (i.e. pre-ring) which is unnatural. You can hear the difference between different sample rates, even when the tweeter doesn't go above 18kHz and/or your hearing stops at 17kHz.

When upsampling CD content, they are using an "apodizing" filter. How did they develop this? Firstly Peter Craven created controlled pre-response anti-alias filters. These have a null near the Nyquist frequency, which will kill any ringing from the original A/D converter.

The ringing is within the time domain that is masked by the human auditory system.

In listening tests, they taught themselves what filter ringing sounded like by starting with a maximum phase filter. This creates a "glassy, bright" sound, and also a kind of phosphorescence or sparkling on top and around the audio.

Then they carried out listening tests on 12 filters. They're not making public all the changes , but the final choice was decided purely by listening tests, and is substantially minimum phase.

Multi-channel audio.

When it came to DVD-A, the recording industry's priorities were copy protection, copy protection, surround sound, and menus.

DVD-A and SACD had a battle, and the iPod won.

Why did it fail? The studios didn't supply content, and CD is already pretty good. Importantly, CDs (which will sound great in great systems) are bought by ordinary people and played in boom boxes - the format is supported by ordinary people. (my interpretation: Any format that isn't, can't survive.)

They have got something into BluRay by stealth - they made the argument that blind people need to be able to operate BluRay player(s), and on this basis the features exist to navigate BluRay without access to a screen - leaving the possibility of using it as an audio-only carrier.

Height is incredibly important - more so than rear channels. He mentioned ADG 2+2+2.

A problem with 5.1 is that 2 good speakers are better than 5 nasty ones. Also, when it comes to laying a room out, height in the front image is more acceptable than placement of rear speakers, but you can't get it from 5.1.

ABX testing

Someone raised the question of the recent blind testing published in the AES that showed 16/44.1 is enough. He replied that, in a familiar system, it's possible to hear things through the system's own faults. However, it can take days to hear a fault.

He argued that Double Blind Testing is used a lot in psychoacoustics, to measure masking etc, but when you move from perception to cognition, it becomes different. ABX testing can disconnect you from things that are continuous. He presented several anti-ABX slides, listing things that DBT proponents believe are inaudible, suggesting that idea that these things sound the same is ridiculous. e.g. 2 violins never sound the same, why would you expect 2 loudspeakers to ever sound the same?

Perceptual memory breaks ABX. He gave the example of listening to a familiar recording on a new high quality audio system: you might hear new details in the recording that you had never appreciated before. He gave an example of a recording where he discovered there were three guitars, rather than the two he had heard before. Having heard this, whenever he listened to the recording afterwards, on almost any system, he could appreciate that there were three guitars.

So in ABX testing, if there's something you can't hear in A, but then you hear it in B, then you'll often be able to hear it when you listen to A again, because your memory fills it in. There was a real difference, but your perceptual memory fills in the gaps to prevent you hearing it more than once.

Hearing is non-linear, and allows you to concentrate on different things, which are both flaws with ABX testing.

(my own note: he didn't state explicitly, but it occurred to me from what was said, that there's a problem when you don't have a break between A and B - because a break tells your brain "this is (or might be) something different", whereas the lack of a break tells you brain "this is more of the same" and may impair the audibility of differences. FWIW I always ABX with breaks between samples, but I'm not sure everyone does - certainly the hardware ABX boxes try to prevent it).

After the official Q&A, I talked to both Bob, and his engineers.

One interesting comment is that the industry wanted HD-DVD or BluRay mainly because China is building DVD players too cheaply, and they need a new format so we can profit from player sales (patents?) again.

The minimum-phase-ish Apodizing filter is used like this:

44.1k > zero pad > 88.2k > apodizing filter @ 22k

In the active loudspeaker cross over filters, they're still using linear phase filters - they sound better. The engineer made the point that even the apodizing filter is still linear phase ish up to ~18kHz.

I asked Bob about his claim that the problem with linear phase brick wall filters is that you hear the wavefront early, and it can damage binaural timing cues. I asked "how can you hear it - it's at 22kHz?!". He said stop thinking about the ear as a linear filter - it's not - it's a wavefront detecting device. I said the cochlear models I'd seen still didn't respond to 22kHz pre-ringing. He said if so, they're missing something - he believed some accurate models would respond. He pointed out that the ear is a very poor performer in many ways, but an excellent difference detector.

I asked how much of the audible difference is due to distortion or non-linearity in the equipment. He said "some, it could be a factor" and then smiled to himself. (my note: I'm not sure how to interpret that smile!). Then he repeated that the ear is a wavefront detector, and ear models which show a simultaneous parallel response should show it.

He also said that he shouldn't say it, but that CD, properly sampled, is pretty much as good as high-res audio. Close enough that it doesn't matter. High Res has slightly better time resolution, but not much.

Talking to his engineers, they said most tests are sighted, or sometimes single-blind. He believes that the A/D anti-alias filter has a bearing on which reconstruction filter sounds best.

Talking to an engineer from another audio company, he said they rarely carry out even single blind tests. They don't believe they're biased because they don't know what to expect to hear - e.g. they change a filter, and it might cause a change in the stereo image, or the change in the sound of a triangle. Different filter = different space perception. While the designer claimed to be able to hear some differences, he said the most subtle changes were only audible to a few golden ears in the company.

That's the lot. If you want it from the horse's mouth, download the mp3 linked in the previous post!

I could add some fairly obvious comments to the ABX section, but I'm sure I don't need to bother.

I could add some fairly obvious comments to the ABX section, but I'm sure I don't need to bother.

It appears to me that he has totally missed the point of ABX testing. ABX can't prove that a difference doesn't exists or can't be heard. It merely tests the hypothesis that a particular listener under specific conditions is able to hear a difference with statistically significant probability.

P.S Thanks for passing on this information, and for all of the contributions you have made to HA over the years.

Thanks for the link. I did listen to the AES UK recording, and recall that he said that 96kHz/24-bit was ludicrously overspecified, and repeated both the sampling rate and bit depth a minute of two later.

That ties in to the 52 kHz sampling rate he later mentioned (implying no more the 26 kHz should be audible to humans at even the loudest level, even though it's unlike anything in music). If that's true, there's 44 kHz of excess sampling frequency (22kHz excess bandwidth) at 96kHz, which is pretty ludicrous. 24-bit is also OTT, though it's often sensible in computers to use an integer number of bytes per sample per channel.

Shannon space seems to encapsulate the 'gamut' of human audition, analogous to the gamut of colour space for human vision.

QUOTE

He mentioned some room noise data from Louis Fielder that's relevant to this discussion.

I hadn't picked up the name, so that's handy if I want to look it up. It sounded like that was from among the quietest listening rooms possible, though the graph he referred to doesn't show up too well on the mp3!

It could well be true but be irrelevant to "real-world" listening rooms with peak level set to about as loud as a human can bear, or ultra-quiet rooms with more modest listening levels. In either of those circumstances, it seems that 16-bit is good enough for the delivery format (though digital reconstruction filters in DACs or digital speakers might as well be implemented in higher resolution to keep any further dither to lower levels.

I thought the beginning was a useful introduction for those not too familiar with speaker design, where he discussed the improvements possible by using active speakers with a separate amplifier dedicated to each driver and the improvements in sensitivity, frequency response, bass extension, loudness, thermal management and protection that could provide compared to traditional passive designs where one has to forego sensitivity, attempt to match all driver impedances and provide power-capable components in the passive crossover.

Someone raised the question of the recent blind testing published in the AES that showed 16/44.1 is enough. He replied that, in a familiar system, it's possible to hear things through the system's own faults. However, it can take days to hear a fault.

He argued that Double Blind Testing is used a lot in psychoacoustics, to measure masking etc, but when you move from perception to cognition, it becomes different. ABX testing can disconnect you from things that are continuous. He presented several anti-ABX slides, listing things that DBT proponents believe are inaudible, suggesting that idea that these things sound the same is ridiculous. e.g. 2 violins never sound the same, why would you expect 2 loudspeakers to ever sound the same?

Perceptual memory breaks ABX. He gave the example of listening to a familiar recording on a new high quality audio system: you might hear new details in the recording that you had never appreciated before. He gave an example of a recording where he discovered there were three guitars, rather than the two he had heard before. Having heard this, whenever he listened to the recording afterwards, on almost any system, he could appreciate that there were three guitars.

So in ABX testing, if there's something you can't hear in A, but then you hear it in B, then you'll often be able to hear it when you listen to A again, because your memory fills it in. There was a real difference, but your perceptual memory fills in the gaps to prevent you hearing it more than once.

Hearing is non-linear, and allows you to concentrate on different things, which are both flaws with ABX testing.

(my own note: he didn't state explicitly, but it occurred to me from what was said, that there's a problem when you don't have a break between A and B - because a break tells your brain "this is (or might be) something different", whereas the lack of a break tells you brain "this is more of the same" and may impair the audibility of differences. FWIW I always ABX with breaks between samples, but I'm not sure everyone does - certainly the hardware ABX boxes try to prevent it).

Actually, the idea that "the hardware ABX boxes try to prevent (breaks between samples) is quite false. If you check the Clark JAES article about ABX you will find that the ABX Comparator schematic published there includes a means for adjusting the break between samples over a wide range. Now some other hardware ABX boxes may not have implemented this feature, but that would be the responsibility of their constructors.

I find it interesting that Mr. Stuart presented the above as being a fact, not a personal theory that needs to be confirmed by testing. That pretty well trashes his Scientist's Card right there! :-(

The second external problem with what he said is that he said his theory breaks just ABX testing. If his theory were correct it would break many forms of testing, including ABC/hr and sighted evaluations. But he didn't say that so that pretty well trashes his backup Scientit's Card.

Basically, we're seing evidence of what may well be Stuart's personal vendetta against ABX. Or, it might be a wish or a dream that he tells himself to reconcile the fact that ABX pretty well rains on many of his favorite parades. If ABX is right, then its hard to escape the idea that Stuart has been spinning his wheels, and his customer's wheels as well.

Now, to get inside Stuart's theory about ABX. The first problem is that Stuart's theory presumes that all ABX tests involve quick switching and that there is always a "lack of a break". This is as false as the similar flase theory that ABX tests all involve short snippets of sound. Anbody who has hands-on experience with ABX testing should know better from personal experience. The switching and the size of the audible samples can be whatever the listener wants them to be.

Another false claim is quoted above:

"He presented several anti-ABX slides, listing things that DBT proponents believe are inaudible, suggesting that idea that these things sound the same is ridiculous. e.g. 2 violins never sound the same, why would you expect 2 loudspeakers to ever sound the same."

I don't believe that there is a widespread belief among ABX proponents that any 2 different loudspeakers sound the same. So, we have a clear case of a straw man argument. I find this level of posturing to be completely dismaying coming from a person of Sturat's stature.

"(Stuart) gave the example of listening to a familiar recording on a new high quality audio system: you might hear new details in the recording that you had never appreciated before. He gave an example of a recording where he discovered there were three guitars, rather than the two he had heard before. Having heard this, whenever he listened to the recording afterwards, on almost any system, he could appreciate that there were three guitars."

This is a fact, but its not just about ABX, its about listener training and it affects just about any kind of listening test you want to do. That a person would hear 3 guitars on a poor system where the sound of 3 guitars might be more masked than on a good one is just an example of listener bias. It is a natural consequence of how human perception works. What it really says is that the idea that listeners should always use music that they are familar with may not always be the best idea.

This is a common method of argumentation that highly biased people use. They come up with some problem that may be quite general, bnut they ascribe it only to something that they want people to distrust. It's like having two girls named Sandy and Julie in a beauty contest. They are both a little zaftig, but someone repatedly says that Sandy is fat. Guess which girl looks at least a little more plump to most people?

Then Stuart further puts his head in the same noose by saying the following:

"Hearing is non-linear, and allows you to concentrate on different things, which are both flaws with ABX testing."

Obviously this possibly real problem (the description is so vague I can't tell whether it is fish or fowl) that he is mentioning potentially afflicts a wide range of listening tests. It might affect all of them.

Bottom line is that there are a lot of inherent problems with just about any kind of listening test. Listening test involve humans, and humans are messy creatures as we are all reminded several times a day. So what are we to do, resolve not to ever do them?

It appears that Bob Stuart is not lighting candles against the darkness created by invalid listening tests but running around and trying to blow out some of the more effective candles that are currently burning.

This report makes it look like Bob Stuart is clearly on the warpath against ABX, and ruining his own reputation for objectivity and honesty by spewing trashy rhetoric, it would seem.

I could add some fairly obvious comments to the ABX section, but I'm sure I don't need to bother.

It appears to me that he has totally missed the point of ABX testing. ABX can't prove that a difference doesn't exists or can't be heard. It merely tests the hypothesis that a particular listener under specific conditions is able to hear a difference with statistically significant probability.

P.S Thanks for passing on this information, and for all of the contributions you have made to HA over the years.